The rapid hybrid acquisition of PN sequences is proposed for DS/CDMA systems. The system introduces the excision CFAR method into the background power estimation. A mathematical analysis is done for the single path and multipath environments. The detection performance of the proposed scheme is compared with that of other acquisition schemes. Results show that the proposed method has better detection performance if the excision coefficient is properly selected.

The present letter deals with the blind equalization problem of a single-input single-output infinite impulse response (SISO-IIR) channel with additive Gaussian noise. To solve the problem, we propose a new criterion for maximizing constrainedly a fourth-order cumulant. The algorithms derived from the criterion have such a novel property that even if Gaussian noise is added to the output of the channel, an effective zero-forcing (ZF) equalizer can be obtained with as little influence of Gaussian noise as possible. To show the validity of the proposed criterion, some simulation results are presented.

In the European Telecommunication Standards Institute (ETSI), Distributed Speech Recognition (DSR) front-end, the distortion added due to feature compression on the front end side increases the variance flooring effect, which in turn increases the identification error rate. The penalty incurred in reducing the bit rate is the degradation in speaker recognition performance. In this paper, we present a nontraditional solution for the previously mentioned problem. To reduce the bit rate, a speech signal is segmented at the client, and the most effective phonemes (determined according to their type and frequency) for speaker recognition are selected and sent to the server. Speaker recognition occurs at the server. Applying this approach to YOHO corpus, we achieved an identification error rate (ER) of 0.05% using an average segment of 20.4% for a testing utterance in a speaker identification task. We also achieved an equal error rate (EER) of 0.42% using an average segment of 15.1% for a testing utterance in a speaker verification task.

We consider the classification problem to construct a classifier c:{0,1}n{0,1} from a given set of examples (training set), which (approximately) realizes the hidden oracle y:{0,1}n{0,1} describing the phenomenon under consideration. For this problem, a number of approaches are already known in computational learning theory; e.g., decision trees, support vector machines (SVM), and iteratively composed features (ICF). The last one, ICF, was proposed in our previous work (Haraguchi et al., (2004)). A feature, composed of a nonempty subset S of other features (including the original data attributes), is a Boolean function fS:{0,1}S{0,1} and is constructed according to the proposed rule. The ICF algorithm iterates generation and selection processes of features, and finally adopts one of the generated features as the classifier, where the generation process may be considered as embodying the idea of boosting, since new features are generated from the available features. In this paper, we generalize a feature to an extended Boolean function fS:{0,1,*}S{0,1,*} to allow partial knowledge, where * denotes the state of uncertainty. We then propose the algorithm ICF* to generate such generalized features. The selection process of ICF* is also different from that of ICF, in that features are selected so as to cover the entire training set. Our computational experiments indicate that ICF* is better than ICF in terms of both classification performance and computation time. Also, it is competitive with other representative learning algorithms such as decision trees and SVM.

The average performance of a single-user MIMO system under spatially correlated fading and with different types of CSI at the transmitter and with perfect CSI at the receiver was studied in recent work. In contrast to analyzing a single performance metric, e.g. the average mutual information or the average bit error rate, we study an arbitrary representative of the class of matrix-monotone functions. Since the average mutual information as well as the average normalized MSE belong to that class, this universal class of performance functions brings together the information theoretic and signal processing performance metric. We use Lowner's representation of operator monotone functions in order to derive the optimum transmission strategies as well as to characterize the impact of correlation on the average performance. Many recent results derived for average mutual information generalize to arbitrary matrix-monotone performance functions, e.g. the optimal transmit strategy without CSI at the transmitter is equal power allocation. The average performance without CSI is a Schur-concave function with respect to transmit and receive correlation. In addition to this, we derive the optimal transmission strategy with long-term statistics knowledge at the transmitter and propose an efficient iterative algorithm. The beamforming-range is the SNR range in which only one data stream spatially multiplexed achieves the maximum average performance. This range is important since it has a simple receiver structure and well known channel coding. We entirely characterize the beamforming-range. Finally, we derive the generalized water-filling transmit strategy for perfect CSI and characterize its properties under channel correlation.

There is still much problem in sizing software despite the existence of well-known software sizing methods such as Function Point method. Many developers still continue to use ad-hoc methods or so called "expert" approaches. This is mainly due to the fact that the existing methods require much information that is difficult to identify or estimate in the early stage of a software project. The accuracy of ad-hoc and "expert" methods also has much problem. The entity-relationship (ER) model is widely used in conceptual modeling (requirements analysis) for data-intensive systems. The characteristic of a data-intensive system, and therefore the source code of its software, is actually well characterized by the ER diagram that models its data. This paper proposes a method for building software size model from extended ER diagram through the use of regression models. We have collected some real data from the industry to do a preliminary validation of the proposed method. The result of the validation is very encouraging.

A simultaneous equations method is one of active noise control algorithms without estimating an error path. This algorithm requires identification of a transfer function from a reference microphone to an error microphone containing the effect of a noise control filter. It is achieved by system identification of an auxiliary filter. However, the introduction of the auxiliary filter requires more number of samples to obtain the noise control filter and brings a requirement of some undesirable assumption in the multiple channel case. In this paper, a new simultaneous equations method without the identification of the auxiliary filter is proposed. By storing a small number of input signals and error signals, we avoid this identification. Therefore, we can reduce the number of samples to obtain the noise control filters and can avoid the undesirable assumption. From simulation examples, it is verified that the merits of the ordinary method is also retained in the proposed method.

This paper investigates the design curves of class E amplifier with nonlinear capacitance for any output Q and finite dc-feed inductance. The important results are; 1) the capacitance nonlinearity strongly affects the design parameters for low Q, 2) the value of dc-feed inductance is hardly affected by the capacitance nonlinearity, and 3) the input voltage is an important parameter to design class E amplifier with nonlinear capacitance. By carrying out PSpice simulations, we show that the simulated results agree with the desired ones quantitatively. It is expected that the design curves in this paper are useful guidelines for the design of class E amplifier with nonlinear capacitance.

We theoretically analyze the performance of coherent ultrashort light pulse code-division multiple-access (CDMA) communication systems with a nonlinear optical thresholder. The coherent ultrashort light pulse CDMA is a promising system for an optical local area network (LAN) due to its advantages of asynchronous transmission, high information security, multiple access capability, and optical processing. The nonlinear optical thresholder is based on frequency chirping induced by self-phase modulation (SPM) in optical fiber, and discriminates an ultrashort pulse from multiple access interference (MAI) with picosecond duration. The numerical results show that the thermal noise caused in a photodetector dominates the bit error rate (BER). BER decreases as the fiber length in the nonlinear thresholder and the photocurrent difference in the photodetector increase. Using the nonlinear optical thresholder allows for the response time of the photodetector to be at least 100 times the duration of the ultrashort pulses. We also show that the optimum cut-off frequency at the nonlinear thresholder to achieve the minimum BER increases with fiber length, the total number of users, and the load resistance in the photodetector.

The Japan Aerospace Exploration Agency (JAXA) plans to launch a geostationary satellite called Engineering Test Satellite VIII (ETS-VIII) in FY 2006. In this paper, a microstrip line array antenna, which has a very simple structure, is introduced to radiate a circularly polarized wave aiming at ETS-VIII applications. This antenna consists of a triangular conducting line with its vertexes rounded off, located above a ground plane, with a gap on one of its side to produce a circular polarization. The proposed antenna is analyzed by numerical simulations for a single element as well as for a three elements array configuration and the possibility of beam-switching in the azimuth space is experimentally confirmed in the latter case. It is found that by properly feeding the elements constituting the array antenna, for an elevation angle El = 48in Tokyo area, three beams are created in the conical-cut direction with a minimum gain more than 6.6 dBic and an axial ratio less than 3 dB.

In this paper, we introduce a new method for depth perception from a 2D natural scene using scale variation of patterns. As the surface from a 2D scene gets farther away from us, the texture appears finer and smoother. Texture gradient is one of the monocular depth cues which can be represented by gradual scale variations of textured patterns. To extract feature vectors from textured patterns, higher order local autocorrelation functions are utilized at each scale step. The hierarchical linear discriminant analysis is employed to classify the scale rate of the feature vector which can be divided into subspaces by recursively grouping the overlapped classes. In the experiment, relative depth perception of 2D natural scenes is performed on the proposed method and it is expected to play an important role in natural scene analysis.

Even though various kinds of IPv6 transition mechanisms have been developed for the transition to an IPv6 network, these transition mechanisms take no stance on whether applications support IPv6 or not. This paper describes why the transition period between IPv4 and IPv6 applications may not be straightforward and applications should be ported to support both IPv4 and IPv6; such applications are called "IP version-independent applications." Also, this paper examines and empirically evaluates overhead of the IP version-independent applications, since the performance implication is not well known. The overhead might be very dependent on data sizes and network performance, but it was relatively minimized for general Internet traffic with larger data sizes and lower network latency.

We consider oscillators consisting of a reactance circuit and a negative resistor. They may happen to have multi-mode oscillations around the anti-resonant frequencies of the reactance circuit. This kind of oscillators can be easily synthesized by setting the resonant and anti-resonant frequencies of the reactance circuits. However, it is not easy to analyze the oscillation phenomena, because they have multiple oscillations whose oscillations depend on the initial guesses. In this paper, we propose a Spice-oriented solution algorithm combining the harmonic balance method with Newton homotopy method that can find out the multiple solutions on the homotopy paths. In our analysis, the determining equations from the harmonic balance method are given by modified equivalent circuit models of "DC," "Cosine" and "Sine" circuits. The modified circuits can be solved by a simulator STC (solution curve tracing circuit), where the multiple oscillations are found by the transient analysis of Spice. Thus, we need not to derive the troublesome circuit equations, nor the mathematical transformations to get the determining equations. It makes the solution algorithms much simpler.

We propose a new receiving method for an information-providing system that uses LED-based traffic lights as the transmitter. We analyzed the improvements obtained when 2-dimentional image sensor replaced the conventional single-element photodiode. First, we discuss the maximum receiver's field of view (FOV) when using the 2-dimentional image sensor at a particular focal length. We analyzed the best vertical inclination for both lanes and quantified the improvements in terms of the enhancement of received signal-noise ratio (SNR) when different numbers of pixels were applied. Our results indicate that using more pixels increases the received SNR and the service area becomes wider compared to the conventional single-element system. Consequently, receivable information within the service area also increased. We also found that the optimum number of pixels to accomplish a reliable communication system is 5050 because performance degradation occured with a larger number of pixels.

Scaling of CMOS Integrated Circuit is becoming difficult, due mainly to rapid increase in power dissipation. How will the semiconductor technology and industry develop? This paper discusses challenges and opportunities in system LSI from three levels of perspectives: transistor level (physics), IC level (electronics), and business level (economics).

Real-time applications are indispensable for conducting research and business in government, industry, and academic organizations. Recently, real-time applications with security requirements increasingly emerged in large-scale distributed systems such as Grids. However, the complexities and specialties of diverse security mechanisms dissuade users from employing existing security services for their applications. To effectively tackle this problem, in this paper we propose a security middleware (SMW) model from which security-sensitive real-time applications are enabled to exploit a variety of security services to enhance the trustworthy executions of the applications. A quality of security control manager (QSCM), a centerpiece of the SMW model, has been designed and implemented to achieve a flexible trade-off between overheads caused by security services and system performance, especially under situations where available resources are dynamically changing and insufficient. A security-aware scheduling mechanism, which plays an important role in QSCM, is capable of maximizing quality of security for real-time applications running in distributed systems as large-scale as Grids. Our empirical studies based on real world traces from a supercomputing center demonstratively show that the proposed model can significantly improve the performance of Grids in terms of both security and schedulability.

A new seamless symbol rate switchable modem for multi-rate FDMA systems is proposed in this paper. In the new modem, a novel clock phase compensation algorithm makes it possible to switch the symbol rate synchronously between the transmitter and the receiver, with no degradation in BER when the symbol rate is changed. In addition, by matching the interpolation filter to the symbol rate, this modem is capable of operating at lower clock speeds, which greatly reduces the consumption power. Computer simulations confirm its fundamental performance. Simulation results show that the proposed power-efficient symbol rate switchable modem can change the symbol rate without degrading BER performance.

We proposed and evaluated a speech packet loss concealment method which predicts lost segments from speech included in packets either before, or both before and after the lost packet. The lost segments are predicted recursively by using linear prediction both in the forward direction from the packet preceding the loss, and in the backward direction from the packet succeeding the lost segment. Predicted samples in each direction are smoothed by averaging using linear weights to obtain the final interpolated signal. The adjacent segments are also smoothed extensively to significantly reduce the speech quality discontinuity between the interpolated signal and the received speech signal. Subjective quality comparisons between the proposed method and the the packet loss concealment algorithm described in the ITU standard G.711 Appendix I showed similar scores up to about 10% packet loss. However, the proposed method showed higher scores above this loss rate, with Mean Opinion Score rating exceeding 2.4, even at an extremely high packet loss rate of 30%. Packet loss concealment of speech degraded with G.729 coding, and babble noise mixed speech showed similar trends, with the proposed method showing higher qualities at high loss rates. We plan to further improve the performance by using adaptive LPC prediction order depending on the estimated pitch, and adaptive LPC bandwidth expansion depending on the consecutive number of repetitive prediction, among many other improvements. We also plan to investigate complexity reduction using gradient LPC coefficient updates, and processing delay reduction using adaptive forward/bidirectional prediction modes depending on the measured packet loss ratio.

This paper discusses the least-squares linear filtering and smoothing (fixed-point and fixed-interval) problems of discrete-time signals from observations, perturbed by additive white noise, which can be randomly delayed by one sampling time. It is assumed that the Bernoulli random variables characterizing delay measurements are correlated in consecutive time instants. The marginal distribution of each of these variables, specified by the probability of a delay in the measurement, as well as their correlation function, are known. Using an innovation approach, the filtering, fixed-point and fixed-interval smoothing recursive algorithms are obtained without requiring the state-space model generating the signal; they use only the covariance functions of the signal and the noise, the delay probabilities and the correlation function of the Bernoulli variables. The algorithms are applied to a particular transmission model with stand-by sensors for the immediate replacement of a failed unit.

We develop several tools to derive quadratic equations from algebraic S-boxes and to prove their linear independence. By applying them to all known almost perfect nonlinear (APN) power functions and the inverse function, we can estimate the resistance against algebraic attacks. As a result, we can show that APN functions have different resistance against algebraic attacks, and especially S-boxes with Gold or Kasami exponents have very weak resistance.